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hexefx_audiolib_F32
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stereo 3 band eq

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pull/2/head
pio 9 months ago
parent ad10d9a088
commit f1f8aa711d
1 changed files with 190 additions and 11 deletions
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  1. 197
      src/filter_3bandeq.h

197
src/filter_3bandeq.h
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@ -25,7 +25,6 @@
#include "mathDSP_F32.h"
#include "basic_components.h"
class AudioFilterEqualizer3band_F32 : public AudioStream_F32
{
public:
@ -33,7 +32,7 @@ public:
{
setBands(500.0f, 3000.0f);
}
void setBands(float32_t bassF, float32_t trebleF)
void setBands(float32_t bassF, float32_t trebleF)
{
trebleF = 2.0f * sinf(M_PI * (trebleF / AUDIO_SAMPLE_RATE_EXACT));
bassF = 2.0f * sinf(M_PI * (bassF / AUDIO_SAMPLE_RATE_EXACT));
@ -66,7 +65,7 @@ public:
{
__disable_irq();
treble_g = t;
mid_g = m;
mid_g = m;
bass_g = b;
__enable_irq();
}
@ -85,7 +84,7 @@ public:
}
for (i = 0; i < block->length; i++)
{
float32_t lpOut, midOut, hpOut; // Low / Mid / High - Sample Values
float32_t lpOut, midOut, hpOut; // Low / Mid / High - Sample Values
float32_t sample = block->data[i]; // give some headroom
// Filter #1 (lowpass)
f1p0 += (lowpass_f * (sample - f1p0)) + vsa;
@ -96,9 +95,9 @@ public:
// // Filter #2 (highpass)
f2p0 += (hipass_f * (sample - f2p0)) + vsa;
f2p1 += (hipass_f * (f2p0 - f2p1));
f2p2 += (hipass_f * (f2p1 - f2p2));
f2p3 += (hipass_f * (f2p2 - f2p3));
f2p1 += (hipass_f * (f2p0 - f2p1));
f2p2 += (hipass_f * (f2p1 - f2p2));
f2p3 += (hipass_f * (f2p2 - f2p3));
hpOut = sdm3 - f2p3;
midOut = sdm3 - (lpOut + hpOut);
@ -123,6 +122,7 @@ public:
bp ^= 1;
return bp;
}
private:
audio_block_f32_t *inputQueueArray[1];
bool bp = false;
@ -137,8 +137,8 @@ private:
float32_t f2p3 = 0.0f;
float32_t sdm1 = 0.0f;
float32_t sdm2 = 0.0f; // 2
float32_t sdm3 = 0.0f; // 3
float32_t sdm2 = 0.0f; // 2
float32_t sdm3 = 0.0f; // 3
static constexpr float32_t vsa = (1.0 / 4294967295.0); // Very small amount (Denormal Fix)
float32_t lpreg;
@ -148,6 +148,185 @@ private:
float32_t hipass_f;
float32_t treble_g = 1.0f;
float32_t mid_g = 1.0f;
};
class AudioFilterEqualizer3bandStereo_F32 : public AudioStream_F32
{
public:
AudioFilterEqualizer3bandStereo_F32(void) : AudioStream_F32(2, inputQueueArray)
{
setBands(500.0f, 3000.0f);
}
void setBands(float32_t bassF, float32_t trebleF)
{
trebleF = 2.0f * sinf(M_PI * (trebleF / AUDIO_SAMPLE_RATE_EXACT));
bassF = 2.0f * sinf(M_PI * (bassF / AUDIO_SAMPLE_RATE_EXACT));
__disable_irq();
lowpass_f = bassF;
hipass_f = trebleF;
__enable_irq();
}
void treble(float32_t t)
{
__disable_irq();
treble_g = t;
__enable_irq();
}
void mid(float32_t m)
{
__disable_irq();
mid_g = m;
__enable_irq();
}
void bass(float32_t b)
{
__disable_irq();
bass_g = b;
__enable_irq();
}
void set(float32_t t, float32_t m, float32_t b)
{
__disable_irq();
treble_g = t;
mid_g = m;
bass_g = b;
__enable_irq();
}
void update()
{
audio_block_f32_t *blockL, *blockR;
int i;
if (bp) // bypass mode
{
blockL = AudioStream_F32::receiveReadOnly_f32(0);
blockR = AudioStream_F32::receiveReadOnly_f32(1);
if (!blockL || !blockR)
{
if (blockL)
AudioStream_F32::release(blockL);
if (blockR)
AudioStream_F32::release(blockR);
return;
}
AudioStream_F32::transmit(blockL, 0);
AudioStream_F32::transmit(blockR, 1);
AudioStream_F32::release(blockL);
AudioStream_F32::release(blockR);
return;
}
blockL = AudioStream_F32::receiveWritable_f32(0);
blockR = AudioStream_F32::receiveWritable_f32(1);
if (!blockL || !blockR)
{
if (blockL)
AudioStream_F32::release(blockL);
if (blockR)
AudioStream_F32::release(blockR);
return;
}
for (i = 0; i < blockL->length; i++)
{
float32_t lpOut, midOut, hpOut; // Low / Mid / High - Sample Values
// channel L
float32_t sample = blockL->data[i]; // give some headroom
// Filter #1 (lowpass)
f1p0[0] += (lowpass_f * (sample - f1p0[0])) + vsa;
f1p1[0] += (lowpass_f * (f1p0[0] - f1p1[0]));
f1p2[0] += (lowpass_f * (f1p1[0] - f1p2[0]));
f1p3[0] += (lowpass_f * (f1p2[0] - f1p3[0]));
lpOut = f1p3[0];
// // Filter #2 (highpass)
f2p0[0] += (hipass_f * (sample - f2p0[0])) + vsa;
f2p1[0] += (hipass_f * (f2p0[0] - f2p1[0]));
f2p2[0] += (hipass_f * (f2p1[0] - f2p2[0]));
f2p3[0] += (hipass_f * (f2p2[0] - f2p3[0]));
hpOut = sdm3[0] - f2p3[0];
midOut = sdm3[0] - (lpOut + hpOut);
// // Scale, Combine and store
lpOut *= bass_g;
midOut *= mid_g;
hpOut *= treble_g;
// // Shuffle history buffer
sdm3[0] = sdm2[0];
sdm2[0] = sdm1[0];
sdm1[0] = sample;
blockL->data[i] = (lpOut + midOut + hpOut);
// channel R
sample = blockR->data[i]; // give some headroom
// Filter #1 (lowpass)
f1p0[1] += (lowpass_f * (sample - f1p0[1])) + vsa;
f1p1[1] += (lowpass_f * (f1p0[1] - f1p1[1]));
f1p2[1] += (lowpass_f * (f1p1[1] - f1p2[1]));
f1p3[1] += (lowpass_f * (f1p2[1] - f1p3[1]));
lpOut = f1p3[1];
// // Filter #2 (highpass)
f2p0[1] += (hipass_f * (sample - f2p0[1])) + vsa;
f2p1[1] += (hipass_f * (f2p0[1] - f2p1[1]));
f2p2[1] += (hipass_f * (f2p1[1] - f2p2[1]));
f2p3[1] += (hipass_f * (f2p2[1] - f2p3[1]));
hpOut = sdm3[1] - f2p3[1];
midOut = sdm3[1] - (lpOut + hpOut);
// // Scale, Combine and store
lpOut *= bass_g;
midOut *= mid_g;
hpOut *= treble_g;
// // Shuffle history buffer
sdm3[1] = sdm2[1];
sdm2[1] = sdm1[1];
sdm1[1] = sample;
blockR->data[i] = (lpOut + midOut + hpOut);
}
AudioStream_F32::transmit(blockL, 0);
AudioStream_F32::transmit(blockR, 1);
AudioStream_F32::release(blockL);
AudioStream_F32::release(blockR);
}
void bypass_set(bool s) { bp = s; }
bool bypass_tgl()
{
bp ^= 1;
return bp;
}
private:
audio_block_f32_t *inputQueueArray[2];
bool bp = false;
float32_t f1p0[2] = {0.0f, 0.0f};
float32_t f1p1[2] = {0.0f, 0.0f};
float32_t f1p2[2] = {0.0f, 0.0f};
float32_t f1p3[2] = {0.0f, 0.0f};
float32_t f2p0[2] = {0.0f, 0.0f};
float32_t f2p1[2] = {0.0f, 0.0f};
float32_t f2p2[2] = {0.0f, 0.0f};
float32_t f2p3[2] = {0.0f, 0.0f};
float32_t sdm1[2] = {0.0f, 0.0f};
float32_t sdm2[2] = {0.0f, 0.0f};
float32_t sdm3[2] = {0.0f, 0.0f};
static constexpr float32_t vsa = (1.0 / 4294967295.0); // Very small amount (Denormal Fix)
float32_t lpreg[2];
float32_t hpreg[2];
float32_t lowpass_f;
float32_t bass_g = 1.0f;
float32_t hipass_f;
float32_t treble_g = 1.0f;
float32_t mid_g = 1.0f;
};
#endif
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